similar to: How to restart * thru phone "when convenient "

Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

Greetings: I did some quick searching of my history of this list, and I tried a quick Google search as well, but perhaps someone on the list can quickly answer this question. I have a very nicely working Asterisk system at home with two Digium X100P FXO cards. When my SIP phones want to dial-out I have them setup to grab the first analog card (Zap/1) with the following extensions.conf segment:

I realize this is a shot in the dark, but I'm trying to find a VoIP provider that offers 402 or 712 area code DID numbers. I'm almost completely convinced that no one offers these area codes (eastern Nebraska, western Iowa), however considering the wide audience of this mailing list I thought this would be a good place to ask. I would prefer a provider that allows for Asterisk use, but I

Dear Asterisk Gurus: Our county is finally ready to begin implementing IP telephony. We intend to use a Cisco router as our PSTN gateway and Asterisk as our soft switch. The plan is to use SIP between the Cisco router and Asterisk. We will have a single PRI T1 connected to the Cisco router for PSTN access. My question is this: Are Cisco routers able to pass caller ID information (from PRI

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten =>

Hi there, here is my attempt to initiate a "restart when convenient" from a software SIP phone. exten => 588,1,Answer exten => 588,2,Wait(1) exten => 588,3,Playback(restart-convenient) exten => 588,4,Wait(1) exten => 588,5,Authenticate(00000) exten => 588,6,System(/usr/sbin/asterisk -rx "restart when convenient") exten => 588,7,Hangup The problem: We

> the default should not be to tell people to run CVS code, > that should only be for people interested in hacking on > the code and trying out bleeding-edge features. I second this motion. While I am not a developer I do notice that most projects tend to take this approach. The CVS is generally for those who want to experiment with the 'bleeding edge', and regular releases of

> -----Original Message----- > From: Jared Smith [mailto:jsmith@drgutah.com] > Sent: Monday, 12 January, 2004 10:41 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Turning a profit (WAS: More words > for Allison) > > > On Mon, 2004-01-12 at 04:49, Alastair Maw wrote: > > Hmmm... I think John's turning a profit... :) > > That was my

> I have placed "ignorepat => 9" in just about every context I > can think of in my extensions.conf, but yet when I dial 9 > from my sip devices, the dialtone is broken. I even tried a > nearly untouched version of samples, and it stil doesn't > work. Is there something somewhere else that needs to be set > to make this work properly, like may in the sip

Hi, today I updated the glibc packages on some CentOS machines. After the Update I checked which services/processes I have to restart "yum -C ps" or "needs-restarting" At the most machines I get no information about necessary restarts, but at two machines a long listing : 1 : /sbin/init 386 : /sbin/udevd-d 659 : /sbin/udevd-d 999 : /usr/sbin/vmtoolsd 1103 : auditd 1128 :

Summary: When I use method="wget" with download.file(), I consistently get a download of the entire file. When I use method="internal", I infrequently get the entire file, but usually get only part of the file. This behavior occurs with .cdf (a weather file format - basically binary) from a UCAR site. I am not sure this is a bug, since it could be some internet

Hi, Thanks for the help. I do see these services coming back up if I reboot the system. They are brought down again when puppet run for the first time. Besides, I have never seen Puppet calling chkconfig service off even in the first run, when the services were in a up state after the system installation. I wonder if I missed something. Any other idea? thanks, -fred On Mon, Aug 29, 2011 at

Hi folks, The question I have is regarding to Service resource on Red Hat systems. I have the following: service { [ "anacron", "atd" ]: ensure => stopped, enable => false, hasrestart => true, hasstatus => true, } It runs fine, disabling the service while the system is up. Debugging it, I noticed that it run the following: debug:

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

Hi, I want to do a scripted 'restart when convenient' on a daily basis. This used to work, but since i've upgraded to Asterisk 11.7 it seems it's never convenient to restart the server. My question: how can i tell *why* it's not convenient to restart the server? It used to be some colleague left the receiver OffHook or something like that, but even when i'm fairly

I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never

For anybody who didn't know there is an article on asterisk in February's Linux Journal. AJ

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

Hi! Yes you're right (for windows), but I found this thread http://www.mail-archive.com/asterisk-users@lists.digium.com/msg10695.html and that works! The first card is connected to a normal Telekom NTBA, the second to an internal PBX. There have to be a possibility to configure multiple ISDN cards (e.g. AVM B1 PCI) through capi.conf. How? Or does chan_capi support only one ISDN-Card?